Eight channels audio on i.MX7 with PCM3168

Toradex Colibri i.MX7Bootlin engineer Alexandre Belloni recently worked on a custom carrier board for a Colibri iMX7 system-on-module from Toradex. This system-on-module obviously uses the i.MX7 ARM processor from Freescale/NXP.

While the module includes an SGTL5000 codec, one of the requirements for that project was to handle up to eight audio channels. The SGTL5000 uses I²S and handles only two channels.


I2S timing diagram from the SGTL5000 datasheet

Thankfully, the i.MX7 has multiple audio interfaces and one is fully available on the SODIMM connector of the Colibri iMX7. A TI PCM3168 was chosen for the carrier board and is connected to the second Synchronous Audio Interface (SAI2) of the i.MX7. This codec can handle up to 8 output channels and 6 input channels. It can take multiple formats as its input but TDM takes the smaller number of signals (4 signals: bit clock, word clock, data input and data output).

TDM timing diagram from the PCM3168 datasheet

The current Linux long term support version is 4.9 and was chosen for this project. It has support for both the i.MX7 SAI (sound/soc/fsl/fsl_sai.c) and the PCM3168 (sound/soc/codecs/pcm3168a.c). That’s two of the three components that are needed, the last one being the driver linking both by describing the topology of the “sound card”. In order to keep the custom code to the minimum, there is an existing generic driver called simple-card (sound/soc/generic/simple-card.c). It is always worth trying to use it unless something really specific prevents that. Using it was as simple as writing the following DT node:

        board_sound {
                compatible = "simple-audio-card";
                simple-audio-card,name = "imx7-pcm3168";
                simple-audio-card,widgets =
                        "Speaker", "Channel1out",
                        "Speaker", "Channel2out",
                        "Speaker", "Channel3out",
                        "Speaker", "Channel4out",
                        "Microphone", "Channel1in",
                        "Microphone", "Channel2in",
                        "Microphone", "Channel3in",
                        "Microphone", "Channel4in";
                simple-audio-card,routing =
                        "Channel1out", "AOUT1L",
                        "Channel2out", "AOUT1R",
                        "Channel3out", "AOUT2L",
                        "Channel4out", "AOUT2R",
                        "Channel1in", "AIN1L",
                        "Channel2in", "AIN1R",
                        "Channel3in", "AIN2L",
                        "Channel4in", "AIN2R";

                simple-audio-card,dai-link@0 {
                        format = "left_j";
                        bitclock-master = <&pcm3168_dac>;
                        frame-master = <&pcm3168_dac>;

                        cpu {
                                sound-dai = <&sai2>;
                                dai-tdm-slot-num = <8>;
                                dai-tdm-slot-width = <32>;

                        pcm3168_dac: codec {
                                sound-dai = <&pcm3168 0>;
                                clocks = <&codec_osc>;

                simple-audio-card,dai-link@2 {
                        format = "left_j";
                        bitclock-master = <&pcm3168_adc>;
                        frame-master = <&pcm3168_adc>;

                        cpu {
                                sound-dai = <&sai2>;
                                dai-tdm-slot-num = <8>;
                                dai-tdm-slot-width = <32>;

                        pcm3168_adc: codec {
                                sound-dai = <&pcm3168 1>;
                                clocks = <&codec_osc>;

There are multiple things of interest:

  • Only 4 input channels and 4 output channels are routed because the carrier board only had that wired.
  • There are two DAI links because the pcm3168 driver exposes inputs and outputs separately
  • As per the PCM3168 datasheet:
    • left justified mode is used
    • dai-tdm-slot-num is set to 8 even though only 4 are actually used
    • dai-tdm-slot-width is set to 32 because the codec takes 24-bit samples but requires 32 clocks per sample (this is solved later in userspace)
    • The codec is master which is usually best regarding clock accuracy, especially since the various SoMs on the market almost never expose the audio clock on the carrier board interface. Here, a crystal was used to clock the PCM3168.

The PCM3168 codec is added under the ecspi3 node as that is where it is connected:

&ecspi3 {
        pcm3168: codec@0 {
                compatible = "ti,pcm3168a";
                reg = <0>;
                spi-max-frequency = <1000000>;
                clocks = <&codec_osc>;
                clock-names = "scki";
                #sound-dai-cells = <1>;
                VDD1-supply = <&reg_module_3v3>;
                VDD2-supply = <&reg_module_3v3>;
                VCCAD1-supply = <&reg_board_5v0>;
                VCCAD2-supply = <&reg_board_5v0>;
                VCCDA1-supply = <&reg_board_5v0>;
                VCCDA2-supply = <&reg_board_5v0>;

#sound-dai-cells is what allows to select between the input and output interfaces.

On top of that, multiple issues had to be fixed:

Finally, an ALSA configuration file (/usr/share/alsa/cards/imx7-pcm3168.conf) was written to ensure samples sent to the card are in the proper format, S32_LE. 24-bit samples will simply have zeroes in the least significant byte. For 32-bit samples, the codec will properly ignore the least significant byte.
Also this describes that the first subdevice is the playback (output) device and the second subdevice is the capture (input) device.

imx7-pcm3168.pcm.default {
	@args [ CARD ]
	@args.CARD {
		type string
	type asym
	playback.pcm {
		type plug
		slave {
			pcm {
				type hw
				card $CARD
				device 0
			format S32_LE
			rate 48000
			channels 4
	capture.pcm {
		type plug
		slave {
			pcm {
				type hw
				card $CARD
				device 1
			format S32_LE
			rate 48000
			channels 4

On top of that, the dmix and dsnoop ALSA plugins can be used to separate channels.

To conclude, this shows that it is possible to easily leverage existing code to integrate an audio codec in a design by simply writing a device tree snippet and maybe an ALSA configuration file if necessary.

Author: Alexandre Belloni

Alexandre is Bootlin's co-owner and COO. Alexandre joined as a kernel and embedded Linux engineer in 2013, and became co-owner and COO in 2021. More details...

9 thoughts on “Eight channels audio on i.MX7 with PCM3168”

  1. Dear Alexandre,

    Thanks for share your valuable experience on this topic.
    please put a schematic connection between codec and iMX7 so the Hardware layer become as clear as the software.

    1. Dear Alexandre,

      Thanks for the wonderfull explanation and I will like to also test this on my own. Can you supply a schematic please?

      1. Unfortunately, I can’t provide schematics. However, there is really not much to it. I that case, the pcm3168 to imx7 connections where the following: LRCKAD to SAI2_RX_SYNC, BCKAD to SAI2_RX_BCLK, DOUT1 to SAI2_RX_DATA0, LRCKDA to SAI2_TX_SYNC, BCKDA to SAI2_TX_BCLK, DIN1 to SAI2_TX_DATA0. The other connection was SPI to control the codec, as shown in the blog post.

  2. Hello Alexandre,
    I am working on similar kind of setup with iMX8M mini board with 4 channel audio interface.
    I did the single stereo channel interface with iMX8M board successfully, but not understanding TDM interface flow. How we can we record the TDM audio stream to iMX8M board.
    e.g. for single channel i used
    arecord -f S16_LE -c2 -r48000 -D hw:1,0 fileName
    And also how to separate those audio channels from TDM audio stream.

    Any reference/ guidance will help.
    Thank you.!

  3. Hello Alexandre,

    thank you for your guidance.
    Can you upload your &sai2 node details for the above configuration, so that i can take some reference from that.

    1. Well, the is not much to that node, it mainly reuses what is provided by the SoC dtsi:

      &iomuxc {
      	pinctrl_sai2: sai2-grp {
      		fsl,pins = <
      			MX7D_PAD_SAI2_TX_BCLK__SAI2_TX_BCLK	0x1f
      			MX7D_PAD_SAI2_TX_SYNC__SAI2_TX_SYNC	0x1f
      			MX7D_PAD_SD2_DATA3__SAI2_TX_DATA0	0x30
      			MX7D_PAD_SD2_DATA0__SAI2_RX_DATA0	0x1f
      			MX7D_PAD_SAI1_RX_BCLK__SAI2_RX_BCLK	0x1f
      			MX7D_PAD_SAI1_RX_SYNC__SAI2_RX_SYNC	0x1f
      &sai2 {
      	pinctrl-names = "default";
      	pinctrl-0 = <&pinctrl_sai2>;
      	status = "okay";
  4. Hello Alexandre,
    Very good article.
    I’m starting a new project, in which this article can help me, but I have some doubts.

    What I intend to develop is a central equipment “A” (with the micro imx8m Mini and the codec pcm3168A) and I want to connect 6 audios from 6 different audio devices “B” (which only contain an analog microphone and a loudspeaker).
    From each audio device I want to be able to communicate (through A) via SIP with a central (telephone) independently.
    I imagine that is possible, no?
    With the IMX8mm, can I have 6 different’s audio interfaces and configure a SIP account for each one and make different calls to each one of them?


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